U.S. patent number 4,755,994 [Application Number 06/773,358] was granted by the patent office on 1988-07-05 for capacity expander for telephone line.
This patent grant is currently assigned to Republic Telcom Systems Corporation. Invention is credited to James M. Kresse, Lester H. Staples.
United States Patent |
4,755,994 |
Staples , et al. |
July 5, 1988 |
Capacity expander for telephone line
Abstract
A private telephone line expander is disclosed which receives
two analog voice inputs. Both input signals are digitized,
compressed using full period splicing, and converted back to analog
for transmission. One is shifted to the upper half of the telephone
line bandwidth. The other remains in the lower half. Both are
transmitted on the same telephone line. When received both signals
are expanded to normal telephone bandwidth and can be connected to
two different telephones. DTMF signals are transmitted with each
voice transmission.
Inventors: |
Staples; Lester H. (Richfield,
MN), Kresse; James M. (Burnsville, MN) |
Assignee: |
Republic Telcom Systems
Corporation (Boulder, CO)
|
Family
ID: |
25097997 |
Appl.
No.: |
06/773,358 |
Filed: |
September 6, 1985 |
Current U.S.
Class: |
370/477; 704/217;
375/240 |
Current CPC
Class: |
H04B
1/66 (20130101); H04Q 5/24 (20130101) |
Current International
Class: |
H04Q
5/00 (20060101); H04B 1/66 (20060101); H04Q
5/24 (20060101); H04J 015/00 (); H04B 001/06 () |
Field of
Search: |
;370/69.1,71,72,118,81
;375/122 ;179/2R,2DP ;381/29,30,31 |
References Cited
[Referenced By]
U.S. Patent Documents
Foreign Patent Documents
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|
|
|
|
|
8400848 |
|
May 1984 |
|
WO |
|
8401315 |
|
Aug 1984 |
|
WO |
|
Other References
Article entitled "A Programmable Voice Digitizer Using the T.I.
TMS-320 Microcomputer", by Daniel F. Daly and L. E. Bergeron, from
Proceedings, IEEE 1983 Acoustics Speech & Signal Processing,
vol. 2..
|
Primary Examiner: Olms; Douglas W.
Assistant Examiner: Scutch, III; Frank M.
Attorney, Agent or Firm: Dorsey & Whitney
Claims
Having described our invention, we claim.
1. A line expander for a telephone line comprising:
means for receiving a first analog voice signal;
means for receiving a second analog voice signal;
autocorrelation means for determining the pitch periods of said
first and second analog voice signals operably coupled to said
means for receiving said analog signals;
voice compression means operably coupled to said autocorrelation
means, for compressing said first and second analog voice signals
based on their respective pitch periods determined by the
autocorrelation means, to form first and second compressed signals
having respective bandwidths less than the respective bandwidths of
said first and second analog voice signals;
audio frequency shifting means for shifting the second compressed
signal to a frequency band above the first compressed signal;
and
means for simultaneously transmitting both compressed voice signals
on the same telephone line.
2. The line expander of claim 1 wherein the compression means
comprises means for splicing successive speech intervals on a
proportional basis.
3. The line expander of claim 1 wherein the expander further
comprises:
means operably coupled to said transmitting means for reducing the
frequency bandwidth of dual tone multiple frequency signals to less
than half of the bandwidth of the telephone line.
4. The private line expander of claim 3 wherein the line expander
further comprises:
means operably coupled to said transmitting means for reducing the
frequency of A, B, C and D tones to less than half of the bandwidth
of the telephone line; and
wherein one of the tones is used for M and E lead signalling and at
least another of the tones is used for diagnostic purposes.
5. A line expander for a private telephone line comprising:
receiving means for receiving compressed analog voice signals;
autocorrelation means, operably coupled to said receiving means,
for determining the pitch of the voice signals; and
means operably coupled to said autocorrelation means for expanding
the analog voice signals based on the pitch determined by the
autocorrelation so that the signals have a bandwidth of normal
audio signals,
said receiving means including means for receiving two or more
compressed analog voice signals on the same telephone line.
6. A line expander for a telephone line comprising:
receiving means for receiving at least two compressed analog voice
signals on the same telephone line;
filter means, operably coupled to said receiving means, for
separating the two compressed voice signals;
first expansion means, operably coupled to said filter means, for
expanding the first compressed voice signal to have a bandwidth of
normal audio frequencies;
audio frequency shifting means, operably coupled to said filter
means for shifting the frequency of the second compressed voice
signal to base band frequencies;
second expansion means operably coupled to said audio frequency
shifting means for expanding the second voice signal to normal
audio frequency; and
first and second phone interface means, respectively coupled to
said first and second expansion means, for connecting each of the
voice signals to separate telephone lines.
7. The line expander of claims 5, or 6 wherein the means for
expanding the analog voice signals comprises means for splicing
successive speech intervals on a proportional basis.
8. The line expander of claims 4, or 5 wherein the private line
expander further comprises:
means for receiving dual tone multiple frequencies associated with
each of the compressed voice signals, each dual tone multiple
frequency being less than half the bandwidth of the telephone line;
and
means for increasing the frequency of each dual tone multiple
frequency received to that normally transmitted on the telephone
line.
9. A line expander for a telephone line comprising:
means for receiving at least one analog voice input;
autocorrelation means for determining the pitch period of the
analog voice input operably coupled to said means for receiving an
analog signal;
voice compression means operably coupled to said autocorrelation
means for compressing the analog voice input to form a compressed
signal based on the pitch period determined by the autocorrelation
means so that the bandwidth of the compressed signal is less than
the bandwidth of the analog voice signal;
means operably coupled to said voice compression means for
transmitting the compressed signal,
said voice compression means including means for splicing
successive speech intervals on a variable proportional basis.
10. The line expander of claim 9 wherein the expander further
comprises:
means for receiving a second analog voice signal;
means for compressing the second voice signal based on the pitch
period of the second voice signal as determined by the
autocorrelation means to form a second compressed signal having a
bandwidth which is less than the bandwidth of the second analog
voice signal;
audio frequency shifting means to shift the second voice signal to
a frequency band above the first compressed signal; and
means for simultaneously transmitting both compressed voice signals
on the same telephone line.
11. A line expander for a telephone line comprising:
means for receiving at least one analog voice input;
autocorrelation means for determining the pitch period of the
analog voice input operably coupled to said means for receiving an
analog signal;
voice compression means operably coupled to said autocorrelation
means, for compressing the analog voice input based on the pitch
period determined by the autocorrelation means to form a compressed
signal so that the bandwidth of the compressed signal is less than
the bandwidth of the analog voice signal;
transmitting means operably coupled to said voice compression means
for transmitting the compressed signal; and
means operably connected to said transmitting means for reducing
the frequency of dual tone multiple frequency signals to less than
half of the bandwidth of the telephone line.
12. The private line expander of claim 11 wherein the line expander
further comprises:
means operably coupled to said transmitting means for reducing the
frequency of A, B, C and D tones to less than half of the bandwidth
of the telephone line; and
wherein one of the tones is used for M and E lead signaling and at
least another of the tones is used for diagnostic purposes.
Description
BACKGROUND OF THE INVENTION
This invention relates to a communication system for transmitting
and receiving information over a telephone line or the like. More
specifically, the invention relates to a device which expands the
capacity of the line by permitting the simultaneous transmission of
two or more voice conversations over the same telephone line or
similar voice grade circuits.
The invention is an improvement over voice signal processing such
as those disclosed in PCT Patent Application No. PCT/US84/00848,
published Dec. 20, 1984 under International Publication No.
WO84/04989 and PCT Patent Application No. PCT/US84/01315, published
on Feb. 28, 1985 under International Publication No.
WO85/00944.
The first patent application identified above discloses an
apparatus for selecting and discarding or duplicating alternate
pitch periods of an analog voice signal.
A peak detector is used to identify glottal pulses in the analog
voice signal. The periodicity of the glottal pulses is used to
determine jump intervals in a microprocessor controlled system
which stores successive samples of the analog voice signal. By
using this method and apparatus alternate pitch periods can be
selected to create a compressed version of the speech signal.
Similarly, the speech signal can be expanded by using jump logic to
duplicate successive speech intervals or glottal epoches
received.
The second patent referenced above, discloses a method of stacking
compressed analog speech signals and related communication signals
so that two speech signals can be transmitted on the same telephone
line.
As shown in the figures to that patent, particularly FIGS. 3 and 4,
stacking is achieved with intermediate frequency techniques by
modulating one of the two analog voice signals with a 455 kilohertz
carrier and then demodulating it with a 451.8 kilohertz carrier
prior to transmission. A similar process is utilized at the
receiving end of the telephone line after which the signal is
expanded for normal audio processing.
SUMMARY OF THE INVENTION
Applicants have devised numerous improvements to the above
captioned inventions using microprocessor based technology which
greatly enhances the operation and performance of the inventions
identified above. Rather than attempting to measure periodicity of
speech intervals with an electronic peak detector detecting glottal
pulses, applicants have devised an autocorrelation scheme, software
based, which accurately detects pitch periods for purposes of
speech compression and expansion. Applicants further, when
combining pitch periods for purposes of compression and when
duplicating pitch periods for purposes of expansion, employ
software techniques to achieve full period splicing. Therefore,
speech when transmitted and when reconstructed for purposes of
expansion flows smoothly from pitch period to pitch period without
anomolies which occur in methods using less than full period
splicing.
Stacking of two or more speech signals is also accomplished at a
substantially reduced cost by stacking at audio frequencies rather
than at intermediate frequencies as suggested by the prior art.
Consequently, audio filters and techniques can be utilized rather
than the more expensive requirements and techniques required at
intermediate frequency ranges.
With the invention two or more simultaneous voice calls can be made
on each private telephone line. The result is an increase in the
capacity of the line or a reduction in the cost of telephone lines.
The unit works with any existing private telephone line and is cost
effective on small trunk groups and on short distance private
lines.
Today, over half the private lines in the United States are in
groups of ten or less and under a hundred miles in distance. Thus
the invention is cost effective for applications requiring more
than one private line and relatively short distances. Future
capacity can be added, one line at a time as it is needed. Because
of its versatility, the invention is appropriate for tail circuits,
small trunk groups to branch facilities or FX/OPX applications.
Since two simultaneous voice conversations share the bandwidth of a
single analog private line, the monthly rental or lease costs for
private line voice networks is reduced. The increase in private
line capacity includes point-to-point tie lines, automatic ring
down circuits, foreign exchange (FX) lines and permits an owner to
resell excess private line capacity.
The compactness of the unit permits it to be installed as a table
top unit or rack mounted in a standard equipment rack.
Both originating side and receiving side connections support E and
M lead signalling.
A unique conversion process takes the incoming analog speech signal
of normal bandwidth and transmits a processed analog signal, which
occupies approximately one-half of the bandwidth of the original
speech, and yet contains the information necessary for a receiving
unit at the far end to reproduce the original analog speech signal
with only moderate degradation. A digital speech processor analyzes
the speech and removes the repetitive patterns inherent and
frequent in human speech. The resulting representation of speech
for each channel is then transmitted using one half of the
bandwidth of an analog line. At the receiving end, a reverse
process takes place as the received speech signals are synthesized
into their original form and connected to two or more standard
three kilohertz analog telephone lines.
DESCRIPTION OF THE DRAWINGS
FIG. 1 is a basic block diagram of the signal flow through the
private line expander.
FIG. 2 is a block diagram of the expansion/compression engine.
FIG. 3 3a, and 3b are a basic block diagram of the components which
make up the private line expander.
FIG. 4, consisting of FIGS. 4a through 4c, is a sketch of speech
signals demonstrating the full splice algorithms for successive
speech intervals in FIG. 4a and resulting in the signal for
transmission of compressed speech intervals as shown in FIG. 4b.
FIG. 4c represents the splicing algorithm used for speech
expansion.
FIG. 5, is a schematic of the dual tone multiple frequency (DTMF)
detector and regenerator.
DESCRIPTION OF THE PREFERRED EMBODIMENT
The invention is a private line expander 12 which allows a
plurality of simultaneous voice conversations to share the
bandwidth of a simple analog private line, thus reducing monthly
costs for private-line voice networks. For ease of explanation the
Description of the Preferred Embodiment will disclose the invention
for simultaneously transmitting two voice conversations on a single
private line. It will be obvious to those skilled in the art that
with appropriate modifications made to the sampling frequencies and
to the stacking frequencies that more than two conversations can be
similarly processed for transmission.
Applications in which the increase in private line capacity is
advantageous include point-to-point tie lines, automatic ring-down
circuits, Foreign Exchange (FX) lines and resale of excess private
line capacity. Benefits achieved with the invention include the
ability of transmitting simultaneous voice calls over a single
private line, a reaction of monthly private line costs, and a
reduction in time required to obtain additional private lines. The
invention works with all existing analog private voice networks. It
is cost effective on small trunked groups and on short distance
private lines. It is a rapid and inexpensive re-configuration which
is expandable on a per-line basis.
Both channel-side and facility-side connections to the invention
are preferably analog four-wire E and M connections. Channel-side
options include two wire and four wire SF. The invention utilizes a
multiplexing scheme based on digital speech processing and time
domain harmonic scaling.
Using the invention incoming analog speech 16, 18 is converted to
digital form. A unique conversion process takes the incoming analog
speech signal of normal bandwidth and transmits a processed analog
signal, which occupies approximately one-half of the bandwidth of
the original speech, and yet contains the information necessary for
a receiving unit at the far end to reproduce the original analog
speech signal with only moderate degradation. A digital speech
processor 20 analyzes the speech and blends repetitive patterns
inherent and frequent in human speech. The resulting representation
of speech for each channel is then transmitted using one half of
the bandwidth of an analog line. At the far end or receiving end
the reverse process takes place, as the received speech signals are
synthesized into their original form and the resulting output 26,28
is connected to two standard three kilohertz analog outputs.
Referring to the figures, operation of the invention can be
understood.
The basic block diagram of the signal flow of the invention is
shown in FIG. 1 which receives inputs 16,18 from each of two
handsets through a PBX and also receives one input telephone line
30 which can carry two or more simultaneous telephone
conversations. Similarly, the output of the line expander 12 is two
separate analog outputs 26,28 to the PBX both of which outputs
26,28 are derived from the single line input 30 to the expander 12
and one output 33 to the telephone line which contains both channel
A and channel B inputs in compressed form. Shown in FIG. 2 is the
expansion/compression engine and shown in FIG. 3 is a basic block
diagram of the circuit elements which achieve the processing shown
in FIG. 1.
FIG. 1 shows how two telephone conversations 16,18 are transmitted
and received simultaneously on a single telephone line 33.
As shown in FIG. 1, the first channel 16 is compressed using a
multiprocessor section 40 configured to function as a
expansion/compression engine as shown in FIG. 2 and is connected to
a stacker 44 to transmit the signal in the frequency range of 200
hertz to 1500 hertz. The second channel 18 is compressed and then
modulated with a fixed audio frequency of 3.38 kilohertz in a
bandshifter 48 to create upper and lower sidebands. The carrier
frequency and the upper sideband are filtered off and the resulting
lower sideband is connected to the stacker 44 to transmit at
frequencies of 1800 hertz to 3200 hertz. The frequency response of
the two simultaneously transmitted analog signals over the passband
of 200 hertz to 3.1 kilohertz are equalized with an equalizer 51
and connected to a telephone interface 53 to transmit
simultaneously on the telephone line 33.
The received signal flow, also shown in FIG. 1, is similar. The two
simultaneously transmitted analog signals are received on the line
30 through a phone interface 54 and the first channel is passed
through a low pass filter 58 which passes frequencies below 1500
hertz and the second channel is passed through a bandpass filter 61
which passes frequencies above 1800 hertz and below 3200 hertz. The
high channel is demodulated with a band shifter 64 to recover the
upper sideband in the 200 hertz to 1500 hertz range and the
resulting signals are expanded with a processor section 40 which
detects pitch periods by autocorrelation of samples obtained at a
3.3 kilohertz sampling rate and duplicates successive speech
intervals by full period splicing to produce data to be read at a
6.6 kilohertz sampling rate to synthesize the original four hundred
to 3000 hertz analog signals. Both of these signals 26,28 are
connected to the phone interface 64,65 of the respective handsets
to which they are directed. Using modulation at audio frequencies
eliminates the extra demodulating requirement set forth in the
prior patent applications discussed and enables the use of audio
technology to reduce the cost of the apparatus.
Referring to the overall block diagram of the system as shown in
FIG. 3, i.e. FIGS. 3a and 3b. Three inputs to the private line
expander are labeled "transmit A" 16 which originates from the
microphone in a first telephone handset. "Transmit B" 18 originates
from the microphone in a second telephone handset and the receive
line 30 is the compressed signal having two simultaneous
conversations in compressed form which is received from the
telephone line. The system output, shown on the right hand side of
FIG. 3, is a signal "receive A" 26 which is connected to the
speaker in the first telephone handset, "transmit line" 33 which is
the two conversations "transmit A" and "transmit B" in compressed
form which are transmitted on the line and "receive B" 28 which is
connected to the speaker in the second telephone handset. Each of
these six signals are passed through bandpass filters and are
sequentially processed with a multiplexer/demultiplexer (not
shown).
A shown in FIGS. 1 and 3, the analog voice signal from each of the
channel A and channel B handsets are digitized, compressed and
transmitted on a single line 33. M & E lead signalling, other
control and supervisory signals or tones A, B, C and D and DTMF
signals for dialing purposes are bypassed and processed separately
with detector and regeneration circuitry 70 as shown in connection
with FIG. 5. The ability to achieve DTMF dialing will be discussed
in connection with FIG. 5.
As shown in FIGS. 2 and 3, the uncompressed signals 16,18 to be
transmitted are passed through a low-pass filter 73 and into an
analog-to-digital conversion circuit 75 with digital samples being
taken at a 6.6 kilohertz sampling rate. The digital values are
stored in an analog-to-digital data buffer 78 which stores the
digital values. These values are processed with a general purpose
microprocessor 80, for example a Motorola 68000, and with the
digital signal processor 20, for example a TMS 32010 manufactured
by Texas Instruments.
The TMS 320 digital signal processor 20 performs the
autocorrelation to determine pitch periods, to be described below,
and utilizes the pitch periods to process the received data and to
perform full period splices. As shown in the drawings, ROM chips
83,85 are used for the 68000 microprocessor program and the TMS 320
program. Additional RAM memory 87 is used for data generated by the
68000 microprocessor 80 and dual ported shared storage 98 is
available for the TMS 320 and the 68000 to share data.
Analog to digital operations with the system use a successive
approximation register chip. Each analog-to-digital conversion
requires a number of iterative passes to generate a digital value
to be stored in the A to D buffer 78. Although shown separately in
the drawings the digital-to-analog circuit 75,78 is functionally
part of the analog-to-digital system 90,93 and the use of these
elements for their respective operations is performed at different
clock times to avoid mutual interference.
Pitch detection is achieved with the digital speech processor 20
which performs autocorrelation. The digital speech processor takes
a block of speech of a given sample length or memory depth, for
example, 140 samples.
Each sample is then cross correlated with every other sample within
a selected pitch range to automatically determine the best
correlation and therefore the pitch of the speech intervals.
This is achieved as follows. For a pitch range of, for example,
twenty to eighty samples, the first sample is multiplied by the
twenty-first sample; the second by the twenty-second and so on
until the sixtieth sample is multiplied by the eightieth. These
products are summed and stored. Then the first sample is multiplied
by the twenty-second sample, the second by the twenty-third, and so
on and again summed and stored. Successive samples are processed
until the first is multiplied by the eighty-first sample, etcetera,
and the sixtieth sample is multiplied by the one hundred fortieth
sample and the products are summed and stored.
The largest sum represents the pitch period of the voice intervals
sampled. It will be the product of the largest positive transitions
added to the product of the largest negative transitions when most
closely aligned during the cross correlation.
If the match occurs at 50 samples, then correlation is established
and the one hundred samples of speech are blended by the digital
speech processor 20 to create 50 data values which are then stored
in the digital-to-analog buffer 90. Since only half the data is
present, the digital-to-analog conversion can take place at a 3.3
kilohertz rate as shown in FIG. 2 which compresses the data for
transmission. The pitch period data is then stored in memory for
use to accomplish full period splicing. Autocorrelation continues,
however, by the digital speech processor 20 with an additional
block of speech starting at the 101st sample for the next 140
samples or similarly selected data block selected for
processing.
The TMS 320 digital speech processor 20 also performs full period
splices to compress successive periods for purposes of
transmission.
As shown in FIG. 4, each of two pitch periods, those matched by
autocorrelation, are blended so that the full ending value of each
of the alternate speech periods corresponds to the value at the end
of the period detected by the autocorrelation routine.
Shown in FIG. 4a are three successive speech intervals, 101,102,103
each representing fifty samples of sampled speech at the 6.6
kilohertz rate. The full period splice is achieved by taking
successive portions of each speech interval 101,102 using the
linear proportions indicated by the lines 105,106 superimposed on
the speech intervals. In other words, the first data value will be
100% of the value of the first interval, Y1, and 0% of the value of
the second interval, Y2; the second will be 98% of the magnitude of
the first interval 101 and 2% of the second 102. The final value
will be 0% of the first speech interval 101 and 100% of the second,
102, Y3. The resulting data as stored in the digital-to-analog
buffer 90 is fifty data samples which smoothly transcends from the
magnitude of the beginning of the first speech interval 101 to the
beginning of the third speech 103 interval and similarly for each
speech interval to be transmitted at the 3.3 kilohertz rate.
Therefore, there is a perfect correlation between the end of the
first reconstructed pitch period and the beginning of the next
pitch period to be transmitted.
It should be understood that the illustrations of FIG. 4, speech
intervals beginning and ending at peak values, is for purposes of
illustration and explanation. The correlation is accurate and
processing is accurately accomplished regardless of where peak
values occur in the speech interval(s) sampled.
The read out of data from the digital-to-analog buffer 90 is at a
3.3 kilohertz rate, therefore, compressing the bandwidth of the
information to a 1.5 kilohertz bandwidth which is then passed
through a filter 110 and connected with the second channel
information which is shifted up in frequency on the same telephone
line for simultaneous transmission.
After digital processing of the channel A voice signal and the
channel B voice signal to compress the signals so that they may
both be transmitted simultaneously on the line, the signals are
stacked with channel A occupying the range 200 hertz to 1.5
kilohertz and channel B occupying the range of 1.8 to 3.1
kilohertz. These signals are transmitted in analog form to the
receive line of the receiving line expander. At the receive end the
compressed signals are separated and expanded to 400 hertz to 3
kilohertz each to produce the "receive A" and "receive B"
signals.
With the differences discussed in connection with FIG. 1, the
receive circuitry and software of FIG. 3 operates in substantially
the same way as discussed in connection with the transmit
circuitry. Each analog signal received is sampled at a 3.3
kilohertz rate. The digital speech processor 20 determines the
pitch with autocorrelation. The pitch period is then used to
regenerate data for expansion of speech intervals with full period
splicing which can be sampled and converted at a 6.6 kilohertz
rate. This regenerates the speech at the four hundred hertz to 3
kilohertz bandwidth.
FIG. 4c demonstrates the algorithm for full period splicing of the
received signals for expansion. Four intervals 201-204 are
represented. To expand the second interval from fifty to one
hundred samples or data values necessary for expansion, the one
hundred samples from the first two intervals 201,202 are averaged
with the one hundred samples from the second two intervals 202,203
using the linear proportions indicated by the lines 205,206
superimposed on the speech intervals 201-203. Similarly the next
hundred expanded speech samples are derived from compressed speech
intervals 202,203 and 204.
FIG. 5 shows the M and E lead, control signal and DTMF detection
and regeneration circuitry 70. Dual tone multifrequency signals
when received by the private line expander 12 are connected through
a DTMF receiver 130 to a control ROM 133, the output of which is
connected to a DTMF generator 136 which is operated at 91% of
normal DTMF frequencies. This allows the frequencies to be narrowed
to be within either the upper or lower band being transmitted. In
other words, the upper tone of 1647 hertz is reduced to below 1500
hertz. On the receive end of the line, the second private line
expander 12 receives the DTMF tones at 91% and regenerates the
tones at 100%. In addition to DTMF dialing purposes control tones
A, B, C and D are also utilized through the tone detector and
regenerator. The A tone is used for E & M, E lead and M lead,
handshaking. The B, C and D tones can be used for diagnostics.
Various methods of echo control may also be advantageously used
with the line expander, 12.
Having described a specific embodiment of our invention it will be
obvious to those skilled in the art that various modifications can
be made and still achieve the objectives of the invention.
Consequently, all such variations and modifications are intended to
be within the scope of the appended claims.
* * * * *